mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
22928 Commits (9c9acc50e65963fd75cbbd8b51a5cd3ceaf9f208)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
adefb772c4 |
Fix hangup cause passthrough regression.
The v1.8 -r369258 change to fix the F and F(x) action logic introduced a
regression in passing the hangup cause from the called channel to the
caller channel.
(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified)
Tested by: rmudgett
........
Merged revisions 371860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 371861 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 371862 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Jonathan Rose
|
3f69a4e34f |
chan_sip: Send 408 on retransmit timeout instead of 603
(closes issue ASTERISK-20124) Reported by: Walter Doekes ........ Merged revisions 371824 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371825 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371845 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
504cfd1070 |
chan_sip: Send a manager event to confirm SIPqualifypeer completes
Prior to this patch, Issuing SIPqualifypeer either resulted in an error or if it succeeded, a few \r\ns. This patch adds a SIPqualifypeerComplete event issued as a response when the command is successfully executed. (closes issue AST-969) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371823 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
44d854938b |
Fix misleading documentation in agents.conf.sample regarding ackcall usage.
The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts ........ Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371789 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371790 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
c81d960ed6 |
Fix incorrect documentation of the MailboxStatus manager command.
The "Waiting" field was misdocumented as reporting the number of messages waiting. In reality, it simply indicated the presence or absence of waiting messages. ........ Merged revisions 371782 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371783 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371784 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
David M. Lee
|
0312bb7a7e |
svn:ignore pjproject bin & output for all platforms.
........ Merged revisions 371753 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371754 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
4b50e3f1ee |
Fix incorrectly documented option in queues.conf
sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts ........ Merged revisions 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371748 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371750 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
David M. Lee
|
05fd2ef0a6 |
Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants.
The original implementations simply wrap pthread functions, which take absolute time as an argument. The spinlock version for systems without those functions treated the argument as a delta. This patch fixes the spinlock version to be consistent with the pthread version. (closes issue ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch uploaded by Egor Gorlin (license 6416) ........ Merged revisions 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371720 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kinsey Moore
|
e13db61695 |
Implement workaround for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes crash when "core show locks" is run. This happens regularly in the testsuite since several tests run "core show locks" to help with debugging. This seems to be a fault with libraries on certain operating systems (notably CentOS 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6. (closes issue ASTERISK-20090) ........ Merged revisions 371690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371691 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371692 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Alec L Davis
|
1295d551f9 |
mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE
........ Merged revisions 371662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371663 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371664 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
96de9ecf30 |
I forgot to add the unit tests for scoped locks earlier today.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371633 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Joshua Colp
|
09b121bb50 |
Add support for call-id logging to chan_motif.
Review: https://reviewboard.asterisk.org/r/2077/ ........ Merged revisions 371619 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
6a539ace84 |
Fix misuses of asprintf throughout the code.
This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371591 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
89a5ff859d |
Add scoped locks to Asterisk.
With the SCOPED_LOCK macro, you can create a variable that locks a specific lock and unlocks the lock when the variable goes out of scope. This is useful for situations where many breaks, continues, returns, or other interruptions would require separate unlock statements. With a scoped lock, these aren't necessary. There are specializations for mutexes, read locks, write locks, ao2 locks, ao2 read locks, ao2 write locks, and channel locks. Each of these is a SCOPED_LOCK at heart though. Review: https://reviewboard.asterisk.org/r/2060 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
db69da3667 |
Use thread-local storage to store pj_thread_descs.
pj_thread_register() takes a parameter of type pj_thread_desc. It was assumed that pj_thread_register either used this item temporarily or made a copy of it. Unfortunately, all it does is keep a pointer to the structure in thread-local storage. This means that if our pj_thread_desc goes out of scope, then pjlib will be referencing bogus data quite often, most commonly on operations involving a pj_mutex_t. In our case, our pj_thread_desc was on the stack and went out of scope very shortly after registering our thread with pjlib. With this change, the pj_thread_desc is stored in thread-local storage so the pointer that pjlib keeps in thread-local storage will reference legitimate memory. (closes issue ASTERISK-20237) reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded by Mark Michelson (license #5049) Tested by Jeremy Pepper ........ Merged revisions 371571 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371572 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
a2068c3db6 |
Ignore recovered zero-length secondary UDPTL packets
In some cases, recovering lost packets using the secondary packet recovery mechanism with UDPTL/T.38 can result in the recovery of zero-length packets. These must be ignored or the frame generated from them can cause segfaults and allocation failures. (closes issue ASTERISK-19762) (closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon) ........ Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371545 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371546 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371547 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
6c8f4f5fff |
Fix for commit r371535
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
ca314fe1e2 |
Apply work-around for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes crash when "core show locks" is run. This happens regularly in the testsuite since several tests run "core show locks" to help with debugging. This seems to be a fault with libraries on certain operating systems (notably CentOS 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6. (issue ASTERISK-20090) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371535 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Matthew Jordan
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f737698654 |
Remove old debug code from http configuration loading
(closes issue ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff uploaded by Andrew Latham (license #5985) ........ Merged revisions 371520 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
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e61cd2f5fc |
Fix typo in JabberSend that looked for '2' instead of '@' in recipient argument
The summary says about all there is to say. (closes issue ASTERISK-20239) Reported by: Gregory Porras ........ Merged revisions 371518 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Matthew Jordan
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78fea20ea7 |
Make the name of the "HangupCauseClear" application consistent
The name of the "HangupCauseClear" application is "HangupCauseClear", not "HangupcauseClear". The incorrect case of 'cause' caused the XML documentation to not register properly. As an aside, this commit message felt very awkward, but I'm not sure how else to note that "X", which has to be "X", was referred to as "x". (closes issue ASTERISK-20253) Reported by: Andrew Latham Patches: hangupcause.diff uploaded by Andrew Latham (license #5985) ........ Merged revisions 371516 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
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294365edd2 |
Update module support level on a variety of modules and compiler options
Some core support modules and compiler options were no longer tagged with a module support level. This patch adds 'core' back to those options. Note that this patch modifies a few of the patches provided by Andrew Latham slightly. res_curl and res_fax are both 'core' supported modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham Tested by: mjordan Patches: astcanary.diff (license #5985) uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew Latham soundsxml.diff (license #5985) uploaded by Andrew Latham ........ Merged revisions 371507 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Matthew Jordan
|
eedab0744b |
Fix memory leak in XML documentation
When formatting documentation fields, the XML documentation parser calls xmldoc_get_formatted. This function allocates a string buffer at the beginning of its routine. Unfortunately, on certain code paths, it also calls xmldoc_string_cleanup, which assumes that it will create the string buffer. The previously allocated string buffer is then leaked by the xmldoc_string_cleanup routine. Now: we don't do that. (closes issue AST-932) Reported by: Alexander Homig ........ Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371491 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371492 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Joshua Colp
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1a95c9a906 |
When a peer registers using WebSocket do not resolve the Contact provided.
(closes issue ASTERISK-20238) Reported by: james.mortensen ........ Merged revisions 371482 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371483 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
|
064c7bd456 |
Add instrumentation to subsystem reloads
When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371437 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371438 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371439 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Russell Bryant
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b8b425971c |
rtp: Ensure defaults are set without rtp.conf.
While building up a new install to test chan_motif, I ran into a failure due to icesupport being disabled. This was due to me not having an rtp.conf. It was intended in the code for it to be enabled by default, but it was only applied if rtp.conf existed. This patch updates res_rtp_asterisk to be consistent in how it handles defaults. A few options didn't have their default values set globally, including icesupport. They are now set and icesupport is enabled by default, even if you do not have an rtp.conf. ........ Merged revisions 371425 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371428 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Joshua Colp
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1f64b85106 |
Add some additional H.264 attributes, "max-smbps" and "max-fps", for passthrough.
(closes issue ASTERISK-20206)
Reported by: ddkprog
Patches:
res_format_attr_h264.c.diff uploaded by ddkprog (license 6008)
........
Merged revisions 371426 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
Terry Wilson
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69dc8e3adb |
Handle integer over/under-flow in ast_parse_args
The strtol family of functions will return *_MIN/*_MAX on overflow. To detect when an overflow has happened, errno must be set to 0 before calling the function, then checked afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371398 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371399 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
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34265d5265 |
Add module reload instrumentation for TEST_FRAMEWORK
This adds AMI events for module reloads when Asterisk is built with TEST_FRAMEWORK enabled and corrects generation of the module load AMI event. (issue PQ-1126) ........ Merged revisions 371393 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371394 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371395 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371396 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Jonathan Rose
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d4879edd8e |
chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header
Previously the pvt SIP_OUTGOING flag was used instead, which will frequently flip during reinvites. (closes issue AST-897) Reported by: Thomas Arimont ........ Merged revisions 371357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371358 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371382 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Jonathan Rose
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70ca2e51a1 |
chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
Under certain conditions, a SIP transaction involving directmedia wouldn't trigger a re-invite because the SDP answer was included in an ACK instead of in a message that we would have triggered the invite with. This patch just queues a source change control frame if the dialog is using directmedia when we find sdp for an ACK. (closes issue AST-913) Reported by: Thomas Arimont ........ Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371338 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371355 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Mark Michelson
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f4a34ee89c |
Fix bug where final queue member would not be removed from memory.
If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas ........ Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371313 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371324 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371325 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Michael L. Young
|
7aac43b4b1 |
Fix Segfault When Registering SIP Over WebSockets
The helper function, get_address_family_filter, in chan_sip for dns resolution by address family was not recognizing the websockets transport and resulting in a null pointer being sent to functions in netsock2, in an attempt to determine if we are bound to ANY address ([::]) or not. This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and SIP_TRANSPORT_WSS which results in a sock address being set properly for use in determining the address family. (closes issue ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven Beisiegel, James Mortensen Patches: asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 371295 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371296 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
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837e00a5cc |
Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction
The other instance of this bug was fixed by jcolp/file in r121496. If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged revisions 371270 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371271 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371272 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
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76d642ff69 |
Add HANGUPCAUSE information to callee channels
This adds HANGUPCAUSE information to called channels so that hangup handlers can, in conjunction with predial dialplan execution, access the hangupcause information when the dialed channel hangs up on a one-to-one basis instead of a many-to-one basis as with HANGUPCAUSE usage on the caller channel. Review: https://reviewboard.asterisk.org/r/2069/ (closes issue ASTERISK-20198) ........ Merged revisions 371258 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
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45c6620d74 |
Add test instrumentation
This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371203 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371227 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Mark Michelson
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5d02d8e016 |
Fix problem where incorrect pointer was checked for nullity.
........ Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371199 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371200 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Matthew Jordan
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9d79ccd3db |
Add UPGRADE-11.txt file; update UPGRADE.txt to reflect Asterisk 12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Richard Mudgett
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e2cd045c7e |
Update CHANGES for private party ID.
........ Merged revisions 371146 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371147 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Mark Michelson
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567b35e547 |
Fix a couple of documentation problems in app_queue.c
* The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts ........ Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371142 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371143 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Matthew Jordan
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938ba94264 |
Remove 10 properties, add 11 properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371134 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Richard Mudgett
|
fb6238899b |
Add private representation of caller, connected and redirecting party ids.
This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
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5ff199d99a |
Fix a comparison that was causing presence tests to fail.
A recent change made it so that device state changes that were not actual "changes" would not get reported to subscribers. The problem was that this inadvertently blocked presence updates as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Alexandr Anikin
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cb525e5c38 |
remove ALREADYGONE flag on ooh323 call data by ooh323_indicate
(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
(closes issue ASTERISK-19308)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-19308.patch
........
Merged revisions 371089 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 371090 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Alexandr Anikin
|
bcc1a0142f |
Send re-register packets by GRQ (gatekeeper request) interval
(close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371061 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Alexandr Anikin
|
c7b2858322 |
restore calling cb functions by timer expire
this was broken in rev 369602 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
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ca481359b9 |
Fix pickup extension channel reference error.
You cannot unref a pointer and then expect to ref it again later. * Fix potential NULL pointer deref if the call pickup search fails. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Alexandr Anikin
|
7709885125 |
Introdue 'ooh323 show gk' cli command that show status of connection
to H.323 Gatekeeper (GkClient state) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371043 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Alexandr Anikin
|
6153acebe8 |
Fix to resend GRQ/RRQ if RRJ (registration reject) is received
(close issue ASTERISK-20094) Patches: ASTERISK-20094.patch ........ Merged revisions 371011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371022 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
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18d5041981 |
Use better libss7 detection test and move libpri compile test.
........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371013 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |